Implement nack_count metric for outbound audio rtp streams.

Bug: webrtc:12510
Change-Id: Ia035885bced3c3d202bb9ffeb88c2556d4830e92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225021
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34444}
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 5d7bc71..62dd53d 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -498,6 +498,8 @@
 
   stats.report_block_datas = std::move(call_stats.report_block_datas);
 
+  stats.nacks_rcvd = call_stats.nacks_rcvd;
+
   return stats;
 }